This invention relates to the field of signal correction, and more particularly to a digital compensation/equalization technique which corrects amplitude and/or phase distortions arising from non-ideal linear system performance.
A linear analog system generally has operational characteristics associated therewith which operate to limit an overall system response, e.g., oscillations in response to a step input, limited frequency response, etc. Such limitations may arise for a variety of reasons, including inaccuracy of components used to implement the system (1% to 5% tolerances on component values), drifting of device operation points, e.g., with changes in temperature and other non-ideal performance of individual system components.
In the past, a number of approaches have been employed in attempts to remove undesirable effects introduced by analog signal processing apparatus. Such approaches have been of both an analog and a digital nature. With respect to the analog approaches, a filter circuit is generally employed having a transfer function selected to compensate for the undesirable effects. Consequently, when the signal from the analog signal apparatus is coupled to the analog filter, the resultant signal therefrom will closely resemble the original signal, with the undesirable effects introduced by the analog signal apparatus reduced. While such an approach has offered varying degrees of success, analog filter circuits suffer from the same limitations associated with analog signal processing apparatus. Consequently, such an approach provides a limited solution to the problem.
In an alternate approach, digital filters have been employed. However, the process to determine the necessary design information for the required digital filter has been cumbersome. In particular, a known signal is first passed through the analog signal apparatus. The results thereof are thereafter transformed to the frequency domain. Once in the frequency domain, a description of a desired filter operation may be obtained by dividing a desired response of the analog signal apparatus by the response observed to the known signal. An inverse Fourier Transform operation is thereafter performed to transfer the results of the division process back into the time domain. The foregoing described process results in a description of a digital filter having a transfer function to effect the removal of undesirable effects of the analog signal apparatus. While such an approach does provide design information with respect to the desired digital filter, the resulting digital filter typically has associated therewith a large number of signal taps. Consequently, a large number of computations are required. Such a resulting filter is undesirable from a practical standpoint.
It is consequently an object of the present invention to achieve a digital signal compensator/equilizer which has a small computational load and a simple adaptive algorithm. It is a further object of the present invention to achieve an implementation structure for a compensator/equalizer which is highly flexible and may consequently accommodate different applications.